[Asterisk-Dev] Re: 16 KHz audio ?

[John Todd]

At 12:49 PM -0800 on 12/18/04, James Cloos wrote:
>  >>>>> "John" == John Todd <jtodd at loligo.com> writes:
>
>John> Perhaps there is some alternate royalty-free standards-based
>John> codec which might also fit the bill?
>
>Speex does have wideband (16kHz @ 16bit sampling) and ultra-wideband
>(32 KHz @ 16 bit) modes.  It shouldn't be too difficult to allow the
>ends to negotiate either of those at the same time as they negotiate
>which sub-codec to use.
>
>-JimC

This sounds like a pretty reasonable idea, then, since Speex is 
already "well-known" to the community.

However, this would imply additional knobs in the dialplan and/or in 
the channel configuration in order to select which Speex variants are 
reasonable.

Digging back in my large bin of "potentially worthless posts to 
-users" I find this:

http://lists.digium.com/pipermail/asterisk-users/2003-October/022384.html

Then, we need to get Speex integrated into the DSP offerings of 
various vendors so that it shows up in desktop devices that have 
speakers integrated into them.  <cough>  This is/was enough of a 
tooth-pulling exercise with iLBC, I suspect Speex will be implemented 
some time after the next ice age.  However, for those of us who use 
IAX-based software running on desktop (Linux/MacOS/Windows) 
environments, this could quickly lead to excellent quality sound if 
the right speakers/microphones/soundcards are utilized.

This would involve those working on the IAX libraries and clients, as 
well as the integration of such data into the "core" of Asterisk. 
More complex than it sounds, perhaps, but probably worth it to be 
able to tout * as a "better-than-toll-quality" solution for 
peer-to-peer voice networking.

JT